Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/webrtcsession_unittest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 235 matching lines @@
   }
 
   cricket::TransportChannel* data_rtp_transport_channel() {
     return rtp_transport_channel(data_channel());
   }
 
   cricket::TransportChannel* data_rtcp_transport_channel() {
     return rtcp_transport_channel(data_channel());
   }
 
+  using webrtc::WebRtcSession::SetAudioPlayout;
+  using webrtc::WebRtcSession::SetAudioSend;
+  using webrtc::WebRtcSession::SetVideoPlayout;
+  using webrtc::WebRtcSession::SetVideoSend;
+
  private:
   cricket::TransportChannel* rtp_transport_channel(cricket::BaseChannel* ch) {
     if (!ch) {
       return nullptr;
     }
     return ch->transport_channel();
   }
 
   cricket::TransportChannel* rtcp_transport_channel(cricket::BaseChannel* ch) {
     if (!ch) {
@@ skipping 3114 matching lines @@
   EXPECT_TRUE((local_offer)->Initialize(offer_str, NULL));
   SetLocalDescriptionOfferExpectError(kBundleWithoutRtcpMux, local_offer);
   JsepSessionDescription* remote_offer =
       new JsepSessionDescription(JsepSessionDescription::kOffer);
   EXPECT_TRUE((remote_offer)->Initialize(offer_str, NULL));
   SetRemoteDescriptionOfferExpectError(kBundleWithoutRtcpMux, remote_offer);
   // Trying unmodified SDP.
   SetLocalDescriptionWithoutError(offer);
 }
 
+TEST_F(WebRtcSessionTest, SetAudioPlayout) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  ASSERT_EQ(1u, channel->recv_streams().size());
+  uint32_t receive_ssrc = channel->recv_streams()[0].first_ssrc();
+  double volume;
+  EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume));
+  EXPECT_EQ(1, volume);
+  session_->SetAudioPlayout(receive_ssrc, false);
+  EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume));
+  EXPECT_EQ(0, volume);
+  session_->SetAudioPlayout(receive_ssrc, true);
+  EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume));
+  EXPECT_EQ(1, volume);
+}
+
+TEST_F(WebRtcSessionTest, SetAudioMaxSendBitrate) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+  EXPECT_EQ(-1, channel->max_bps());
+  webrtc::RtpParameters params = session_->GetAudioRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_TRUE(session_->SetAudioRtpSendParameters(send_ssrc, params));
+
+  // Read back the parameters and verify they have been changed.
+  params = session_->GetAudioRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the audio channel received the new parameters.
+  params = channel->GetRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the global bitrate limit has not been changed.
+  EXPECT_EQ(-1, channel->max_bps());
+}
+
+TEST_F(WebRtcSessionTest, SetAudioSend) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  ASSERT_EQ(1u, channel->send_streams().size());
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+  EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
+
+  cricket::AudioOptions options;
+  options.echo_cancellation = rtc::Optional<bool>(true);
+
+  std::unique_ptr<FakeAudioSource> source(new FakeAudioSource());
+  session_->SetAudioSend(send_ssrc, false, options, source.get());
+  EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
+  EXPECT_EQ(rtc::Optional<bool>(), channel->options().echo_cancellation);
+  EXPECT_TRUE(source->sink() != nullptr);
+
+  // This will trigger SetSink(nullptr) to the |source|.
+  session_->SetAudioSend(send_ssrc, true, options, nullptr);
+  EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
+  EXPECT_EQ(rtc::Optional<bool>(true), channel->options().echo_cancellation);
+  EXPECT_TRUE(source->sink() == nullptr);
+}
+
+TEST_F(WebRtcSessionTest, AudioSourceForLocalStream) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  ASSERT_EQ(1u, channel->send_streams().size());
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+
+  std::unique_ptr<FakeAudioSource> source(new FakeAudioSource());
+  cricket::AudioOptions options;
+  session_->SetAudioSend(send_ssrc, true, options, source.get());
+  EXPECT_TRUE(source->sink() != nullptr);
+
+  // Delete the |source| and it will trigger OnClose() to the sink, and this
+  // will invalidate the |source_| pointer in the sink and prevent getting a
+  // SetSink(nullptr) callback afterwards.
+  source.reset();
+
+  // This will trigger SetSink(nullptr) if no OnClose() callback.
+  session_->SetAudioSend(send_ssrc, true, options, nullptr);
+}
+
+TEST_F(WebRtcSessionTest, SetVideoPlayout) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  ASSERT_LT(0u, channel->sinks().size());
+  EXPECT_TRUE(channel->sinks().begin()->second == NULL);
+  ASSERT_EQ(1u, channel->recv_streams().size());
+  uint32_t receive_ssrc = channel->recv_streams()[0].first_ssrc();
+  cricket::FakeVideoRenderer renderer;
+  session_->SetVideoPlayout(receive_ssrc, true, &renderer);
+  EXPECT_TRUE(channel->sinks().begin()->second == &renderer);
+  session_->SetVideoPlayout(receive_ssrc, false, &renderer);
+  EXPECT_TRUE(channel->sinks().begin()->second == NULL);
+}
+
+TEST_F(WebRtcSessionTest, SetVideoMaxSendBitrate) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+  EXPECT_EQ(-1, channel->max_bps());
+  webrtc::RtpParameters params = session_->GetVideoRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_TRUE(session_->SetVideoRtpSendParameters(send_ssrc, params));
+
+  // Read back the parameters and verify they have been changed.
+  params = session_->GetVideoRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the video channel received the new parameters.
+  params = channel->GetRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the global bitrate limit has not been changed.
+  EXPECT_EQ(-1, channel->max_bps());
+}
+
+TEST_F(WebRtcSessionTest, SetVideoSend) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  ASSERT_EQ(1u, channel->send_streams().size());
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+  EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
+  cricket::VideoOptions* options = NULL;
+  session_->SetVideoSend(send_ssrc, false, options, nullptr);
+  EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
+  session_->SetVideoSend(send_ssrc, true, options, nullptr);
+  EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
+}
+
 TEST_F(WebRtcSessionTest, CanNotInsertDtmf) {
   TestCanInsertDtmf(false);
 }
 
 TEST_F(WebRtcSessionTest, CanInsertDtmf) {
   TestCanInsertDtmf(true);
 }
 
 TEST_F(WebRtcSessionTest, InsertDtmf) {
   // Setup
@@ skipping 857 matching lines @@
 }
 
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.
 
 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
                         WebRtcSessionTest,
                         testing::Values(ALREADY_GENERATED,
                                         DTLS_IDENTITY_STORE));